mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-08 18:39:54 +00:00
webrtcbin: Add unit test for closing of data channels
Add test for verifying that the data channel "close" action signal triggers an SCTP_RESET_STREAMS request that is propagated to the other side and eventually leads to both sides closing properly. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
This commit is contained in:
parent
4d514abfd6
commit
caefc3a831
1 changed files with 178 additions and 0 deletions
|
@ -2045,6 +2045,183 @@ GST_START_TEST (test_data_channel_create_after_negotiate)
|
|||
|
||||
GST_END_TEST;
|
||||
|
||||
struct test_data_channel
|
||||
{
|
||||
GObject *dc1;
|
||||
GObject *dc2;
|
||||
gint n_open;
|
||||
gint n_closed;
|
||||
gint n_destroyed;
|
||||
};
|
||||
|
||||
static void
|
||||
have_data_channel_mark_open (struct test_webrtc *t,
|
||||
GstElement * element, GObject * our, gpointer user_data)
|
||||
{
|
||||
struct test_data_channel *tdc = t->data_channel_data;
|
||||
|
||||
tdc->dc2 = our;
|
||||
if (g_atomic_int_add (&tdc->n_open, 1) == 1) {
|
||||
test_webrtc_signal_state_unlocked (t, STATE_CUSTOM);
|
||||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
is_data_channel_open (GObject * channel)
|
||||
{
|
||||
GstWebRTCDataChannelState ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED;
|
||||
|
||||
if (channel) {
|
||||
g_object_get (channel, "ready-state", &ready_state, NULL);
|
||||
}
|
||||
|
||||
return ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN;
|
||||
}
|
||||
|
||||
static void
|
||||
on_data_channel_open (GObject * channel, GParamSpec * pspec,
|
||||
struct test_webrtc *t)
|
||||
{
|
||||
struct test_data_channel *tdc = t->data_channel_data;
|
||||
|
||||
if (is_data_channel_open (channel)) {
|
||||
if (g_atomic_int_add (&tdc->n_open, 1) == 1) {
|
||||
test_webrtc_signal_state (t, STATE_CUSTOM);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
on_data_channel_close (GObject * channel, GParamSpec * pspec,
|
||||
struct test_webrtc *t)
|
||||
{
|
||||
struct test_data_channel *tdc = t->data_channel_data;
|
||||
GstWebRTCDataChannelState ready_state;
|
||||
|
||||
g_object_get (channel, "ready-state", &ready_state, NULL);
|
||||
|
||||
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
|
||||
g_atomic_int_add (&tdc->n_closed, 1);
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
on_data_channel_destroyed (gpointer data, GObject * where_the_object_was)
|
||||
{
|
||||
struct test_webrtc *t = data;
|
||||
struct test_data_channel *tdc = t->data_channel_data;
|
||||
|
||||
if (where_the_object_was == tdc->dc1) {
|
||||
tdc->dc1 = NULL;
|
||||
} else if (where_the_object_was == tdc->dc2) {
|
||||
tdc->dc2 = NULL;
|
||||
}
|
||||
|
||||
if (g_atomic_int_add (&tdc->n_destroyed, 1) == 1) {
|
||||
test_webrtc_signal_state (t, STATE_CUSTOM);
|
||||
}
|
||||
}
|
||||
|
||||
GST_START_TEST (test_data_channel_close)
|
||||
{
|
||||
#define NUM_CHANNELS 3
|
||||
struct test_webrtc *t = test_webrtc_new ();
|
||||
struct test_data_channel tdc = { NULL, NULL, 0, 0, 0 };
|
||||
guint channel_id[NUM_CHANNELS] = { 0, 1, 2 };
|
||||
gulong sigid = 0;
|
||||
int i;
|
||||
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
|
||||
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
|
||||
|
||||
t->on_negotiation_needed = NULL;
|
||||
t->on_ice_candidate = NULL;
|
||||
t->on_data_channel = have_data_channel_mark_open;
|
||||
t->data_channel_data = &tdc;
|
||||
|
||||
fail_if (gst_element_set_state (t->webrtc1,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
fail_if (gst_element_set_state (t->webrtc2,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
|
||||
/* open and close NUM_CHANNELS data channels to verify that we can reuse the
|
||||
* stream id of a previously closed data channel and that we have the same
|
||||
* behaviour no matter if we create the channel in READY or PLAYING state */
|
||||
for (i = 0; i < NUM_CHANNELS; i++) {
|
||||
tdc.n_open = 0;
|
||||
tdc.n_closed = 0;
|
||||
tdc.n_destroyed = 0;
|
||||
|
||||
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
|
||||
&tdc.dc1);
|
||||
g_assert_nonnull (tdc.dc1);
|
||||
g_object_unref (tdc.dc1); /* webrtcbin should still hold a ref */
|
||||
g_object_weak_ref (tdc.dc1, on_data_channel_destroyed, t);
|
||||
g_signal_connect (tdc.dc1, "on-error",
|
||||
G_CALLBACK (on_channel_error_not_reached), NULL);
|
||||
sigid = g_signal_connect (tdc.dc1, "notify::ready-state",
|
||||
G_CALLBACK (on_data_channel_open), t);
|
||||
|
||||
if (i == 0) {
|
||||
fail_if (gst_element_set_state (t->webrtc1,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
|
||||
fail_if (gst_element_set_state (t->webrtc2,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
|
||||
|
||||
test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
|
||||
} else {
|
||||
/* FIXME: Creating a data channel may result in "on-open" being sent
|
||||
* before we even had a chance to register the signal. For this test we
|
||||
* want to make sure that the channel is actually open before we try to
|
||||
* close it. So if we didn't receive the signal we fall back to a 1s
|
||||
* timeout where we explicitly check if both channels are open. */
|
||||
gint64 timeout = g_get_monotonic_time () + 1 * G_TIME_SPAN_SECOND;
|
||||
g_mutex_lock (&t->lock);
|
||||
while (((1 << t->state) & STATE_CUSTOM) == 0) {
|
||||
if (!g_cond_wait_until (&t->cond, &t->lock, timeout)) {
|
||||
g_assert (is_data_channel_open (tdc.dc1)
|
||||
&& is_data_channel_open (tdc.dc2));
|
||||
break;
|
||||
}
|
||||
}
|
||||
g_mutex_unlock (&t->lock);
|
||||
}
|
||||
|
||||
g_object_get (tdc.dc1, "id", &channel_id[i], NULL);
|
||||
|
||||
g_signal_handler_disconnect (tdc.dc1, sigid);
|
||||
g_object_weak_ref (tdc.dc2, on_data_channel_destroyed, t);
|
||||
g_signal_connect (tdc.dc1, "notify::ready-state",
|
||||
G_CALLBACK (on_data_channel_close), t);
|
||||
g_signal_connect (tdc.dc2, "notify::ready-state",
|
||||
G_CALLBACK (on_data_channel_close), t);
|
||||
test_webrtc_signal_state (t, STATE_NEW);
|
||||
|
||||
/* currently we assume there is no renegotiation if the last data channel is
|
||||
* removed but if it changes this test could be extended to verify both
|
||||
* the behaviour of removing the last channel as well as the behaviour when
|
||||
* there are still data channels remaining */
|
||||
t->on_negotiation_needed = _negotiation_not_reached;
|
||||
g_signal_emit_by_name (tdc.dc1, "close");
|
||||
|
||||
test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
|
||||
|
||||
assert_equals_int (g_atomic_int_get (&tdc.n_closed), 2);
|
||||
assert_equals_pointer (tdc.dc1, NULL);
|
||||
assert_equals_pointer (tdc.dc2, NULL);
|
||||
|
||||
test_webrtc_signal_state (t, STATE_NEW);
|
||||
}
|
||||
|
||||
/* verify the same stream id has been reused for each data channel */
|
||||
assert_equals_int (channel_id[0], channel_id[1]);
|
||||
assert_equals_int (channel_id[0], channel_id[2]);
|
||||
|
||||
test_webrtc_free (t);
|
||||
#undef NUM_CHANNELS
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
static void
|
||||
on_buffered_amount_low_emitted (GObject * channel, struct test_webrtc *t)
|
||||
{
|
||||
|
@ -3752,6 +3929,7 @@ webrtcbin_suite (void)
|
|||
tcase_add_test (tc, test_data_channel_transfer_string);
|
||||
tcase_add_test (tc, test_data_channel_transfer_data);
|
||||
tcase_add_test (tc, test_data_channel_create_after_negotiate);
|
||||
tcase_add_test (tc, test_data_channel_close);
|
||||
tcase_add_test (tc, test_data_channel_low_threshold);
|
||||
tcase_add_test (tc, test_data_channel_max_message_size);
|
||||
tcase_add_test (tc, test_data_channel_pre_negotiated);
|
||||
|
|
Loading…
Reference in a new issue